SIP/2.0 400 Invalid Contact information

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  Hi Scott,

   

  Try to have the media gateway present the From number as E.164 with a + prefixed to the Mediation Server. Also make sure you have configured a location profile on the Mediation Server and related normalization rules and have configured your user correctly with msRTCSIP-Line. Have a look in the OCS_VoIP_Guide.

   

  Also I don't think your media gateway is on the list of qualified media gateways http://technet.microsoft.com/en-us/office/bb735838.aspx, so you might run into more problems.

   

  best regards

     Jens

  Thursday, November 29, 2007 10:34 AM
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  I believe I have set up the system correctly, but it still does not work. I now send the + in the FROM field.

   

  TL_INFO(TF_PROTOCOL) [0]09C4.0ACC::11/29/2007-11:50:12.430.00000b65 (S4,SipMessage.DataLoggingHelper:472.idx(550))
  <<<<<<<<<<<<Incoming SipMessage c=[<SipTcpConnection_8B4A0E>], 192.168.195.114:5060<-192.168.195.190:1087
  INVITE sip:+4852@192.168.195.114:5060;transport=tcp SIP/2.0
  FROM: <sip:+33630704615@192.168.195.114>;tag=q-18be-ee5d
  TO: <sip:+4852@192.168.195.114>
  CSEQ: 1 INVITE
  CALL-ID: 12840810511191070@192.168.195.190
  MAX-FORWARDS: 70
  VIA: SIP/2.0/TCP 192.168.195.190:5060;branch=z9hG4bKfrbfjbdC001543211
  RECORD-ROUTE: <sip:192.168.195.190>
  CONTACT: <sip:+33630704615@192.168.195.190;transport=tcp>
  CONTENT-LENGTH: 253
  USER-AGENT: QuesCom SIP Gateway 5.00.007
  CONTENT-TYPE: application/sdp
  ALLOW: INVITE, BYE, ACK, CANCEL, REGISTER, OPTIONS, REFER, NOTIFY, INFO
  v=0
  o=QuesCom 18467 18467 IN IP4 192.168.195.190
  s=NonSIP
  c=IN IP4 192.168.195.190
  t=0 0
  m=audio 11010 RTP/AVP 8 0 18 101
  a=rtpmap:8 pcma/8000/1
  a=rtpmap:0 pcmu/8000/1
  a=rtpmap:18 g729/8000/1
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-15
  ------------EndOfIncoming SipMessage
  TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-11:50:12.440.00000b71 (S4,SipMessage.DataLoggingHelper:472.idx(500))
  >>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_8B4A0E>], 192.168.195.114:5060->192.168.195.190:1087
  SIP/2.0 100 Trying
  FROM: <sip:+33630704615@192.168.195.114>;tag=q-18be-ee5d
  TO: <sip:+4852@192.168.195.114>
  CSEQ: 1 INVITE
  CALL-ID: 12840810511191070@192.168.195.190
  VIA: SIP/2.0/TCP 192.168.195.190:5060;branch=z9hG4bKfrbfjbdC001543211
  CONTENT-LENGTH: 0
  ------------EndOfOutgoing SipMessage
  TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-11:50:12.530.00000bff (S4,SipMessage.DataLoggingHelper:472.idx(500))
  >>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_3300B4D>], 192.168.195.114:3684->192.168.195.113:5061
  INVITE sip:+4852@testlcs.local;user=phone SIP/2.0
  FROM: <sip:+33630704615@testlcs.local;user=phone>;epid=50FADD4DA6;tag=e8952cc14
  TO: <sip:+4852@testlcs.local;user=phone>
  CSEQ: 21 INVITE
  CALL-ID: 5e80aad3-8fca-4f90-bc1b-db6a9baf35bf
  MAX-FORWARDS: 70
  VIA: SIP/2.0/TLS 192.168.195.114:3684;branch=z9hG4bK571d6d76
  CONTACT: <sipcottlcs2007med.testlcs.local@testlcs.local;gruu;opaque=srvr:MediationServer:mc3q_s6qrk-5EezN3UoChgAA;grid=c60f17471e554939854cdcd79523abfc>;isGateway
  CONTENT-LENGTH: 944
  SUPPORTED: replaces
  SUPPORTED: gruu-10
  USER-AGENT: RTCC/3.0.0.0 MediationServer
  CONTENT-TYPE: application/sdp; charset=utf-8
  ALLOW: UPDATE
  ms-call-source: non-ms-rtc
  ALLOW: Ack, Cancel, Bye,Invite,Refer
  v=0
  o=- 0 0 IN IP4 192.168.195.114
  s=session
  c=IN IP4 192.168.195.114
  b=CT:1000
  t=0 0
  m=audio 61355 RTP/AVP 97 101 115 111 0 8
  c=IN IP4 192.168.195.114
  a=rtcp:62123
  a=candidatevSSEml9F2cnJP93SGZ8bu7msFdcHmm80vSLWZTdVLM 1 omXUt93kRdTdCzHCq9D+9A UDP 0.900 192.168.195.114 61355
  a=candidatevSSEml9F2cnJP93SGZ8bu7msFdcHmm80vSLWZTdVLM 2 omXUt93kRdTdCzHCq9D+9A UDP 0.900 192.168.195.114 62123
  a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 inline:jJPJeA1RC89pyUK/XPtcyytVP84gRDi5rCgQTq7K|2^31|1:1
  a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:cPvwZ5OF9N9zutdg2JoQIV6XtFpuYULob027AB7w|2^31|1:1
  a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:QjWjKn1QvQIPfhj+ofhp93006LYhIsb6sazBwwjy|2^31
  a=label:main-audio
  a=rtpmap:97 RED/8000
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-16
  a=rtpmap:115 x-msrta/8000
  a=fmtp:115 bitrate=11800
  a=rtpmap:111 SIREN/16000
  a=fmtp:111 bitrate=16000
  a=rtpmap:0 PCMU/8000
  a=rtpmap:8 PCMA/8000
  a=ptime:20
  ------------EndOfOutgoing SipMessage
  TL_INFO(TF_PROTOCOL) [0]09C4.0ACC::11/29/2007-11:50:14.393.00000c24 (S4,SipMessage.DataLoggingHelper:472.idx(550))
  <<<<<<<<<<<<Incoming SipMessage c=[<SipTlsConnection_3300B4D>], 192.168.195.114:3684<-192.168.195.113:5061
  SIP/2.0 400 Invalid Contact information
  FROM: <sip:+33630704615@testlcs.local;user=phone>;epid=50FADD4DA6;tag=e8952cc14
  TO: <sip:+4852@testlcs.local;user=phone>;tag=FE355C133B228334B697EF1833317D62
  CSEQ: 21 INVITE
  CALL-ID: 5e80aad3-8fca-4f90-bc1b-db6a9baf35bf
  VIA: SIP/2.0/TLS 192.168.195.114:3684;branch=z9hG4bK571d6d76;ms-received-port=3684;ms-received-cid=29E00
  CONTENT-LENGTH: 0
  ms-diagnostics: 1018;reason="Parsing failure";source="scottlcs2007.testlcs.local"
  ------------EndOfIncoming SipMessage
  TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-11:50:14.393.00000c33 (S4,SipMessage.DataLoggingHelper:472.idx(500))
  >>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_3300B4D>], 192.168.195.114:3684->192.168.195.113:5061
  ACK sip:+4852@testlcs.local;user=phone SIP/2.0
  FROM: <sip:+33630704615@testlcs.local;user=phone>;tag=e8952cc14;epid=50FADD4DA6
  TO: <sip:+4852@testlcs.local;user=phone>;tag=FE355C133B228334B697EF1833317D62
  CSEQ: 21 ACK
  CALL-ID: 5e80aad3-8fca-4f90-bc1b-db6a9baf35bf
  MAX-FORWARDS: 70
  VIA: SIP/2.0/TLS 192.168.195.114:3684;branch=z9hG4bK571d6d76
  CONTENT-LENGTH: 0
  ------------EndOfOutgoing SipMessage
  TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-11:50:14.503.00000c5b (S4,SipMessage.DataLoggingHelper:472.idx(500))
  >>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_8B4A0E>], 192.168.195.114:5060->192.168.195.190:1087
  SIP/2.0 400 Invalid Contact information
  FROM: <sip:+33630704615@192.168.195.114>;tag=q-18be-ee5d
  TO: <sip:+4852@192.168.195.114>;tag=1ead26a45
  CSEQ: 1 INVITE
  CALL-ID: 12840810511191070@192.168.195.190
  VIA: SIP/2.0/TCP 192.168.195.190:5060;branch=z9hG4bKfrbfjbdC001543211
  CONTENT-LENGTH: 0
  SERVER: RTCC/3.0.0.0 MediationServer
  ------------EndOfOutgoing SipMessage
  TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-11:50:14.513.00000cca (S4,SipMessage.DataLoggingHelper:472.idx(500))
  >>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_1B76E14>], 192.168.195.114:3685->192.168.195.113:5061
  SERVICE sip:+4852@testlcs.local;user=phone SIP/2.0
  FROM: <sip:+4852@testlcs.local;user=phone>;epid=50FADD4DA6;tag=89af6120c0
  TO: <sip:+4852@testlcs.local;user=phone>
  CSEQ: 22 SERVICE
  CALL-ID: 9454d7a1041648d0a2f0f229299ff1f2
  MAX-FORWARDS: 70
  VIA: SIP/2.0/TLS 192.168.195.114:3685;branch=z9hG4bKf4b48eb2
  CONTENT-LENGTH: 493
  USER-AGENT: RTCC/3.0.0.0 MediationServer
  CONTENT-TYPE: application/msrtc-reporterror+xml
  <?xml version="1.0" encoding="us-ascii"?><reportError xmlns="http://schemas.microsoft.com/2006/09/sip/error-reporting"><error callId="5e80aad3-8fca-4f90-bc1b-db6a9baf35bf" toUri="sip:+4852@testlcs.local;user=phone" fromTag="e8952cc14" toTag="FE355C133B228334B697EF1833317D62" requestType="INVITE" contentType="application/sdp;call-type=audio" responseCode="400"><diagHeader>1018;reason="Parsing failure";source="scottlcs2007.testlcs.local"</diagHeader><progressReports /></error></reportError>------------EndOfOutgoing SipMessage
  TL_INFO(TF_PROTOCOL) [0]09C4.0ACC::11/29/2007-11:50:16.796.00000cf3 (S4,SipMessage.DataLoggingHelper:472.idx(550))
  <<<<<<<<<<<<Incoming SipMessage c=[<SipTlsConnection_1B76E14>], 192.168.195.114:3685<-192.168.195.113:5061
  SIP/2.0 401 Unauthorized
  FROM: <sip:+4852@testlcs.local;user=phone>;epid=50FADD4DA6;tag=89af6120c0
  TO: <sip:+4852@testlcs.local;user=phone>;tag=FE355C133B228334B697EF1833317D62
  CSEQ: 22 SERVICE
  CALL-ID: 9454d7a1041648d0a2f0f229299ff1f2
  VIA: SIP/2.0/TLS 192.168.195.114:3685;branch=z9hG4bKf4b48eb2;ms-received-port=3685;ms-received-cid=29F00
  WWW-AUTHENTICATE: NTLM realm="SIP Communications Service", targetname="scottlcs2007.testlcs.local", version=3
  WWW-AUTHENTICATE: Kerberos realm="SIP Communications Service", targetname="sip/scottlcs2007.testlcs.local", version=3
  CONTENT-LENGTH: 0
  DATE: Thu, 29 Nov 2007 11:50:14 GMT
  ------------EndOfIncoming SipMessage
  

  Thanks again for your help.

  Scott

   

  Thursday, November 29, 2007 1:25 PM
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  Same problem on my OCS <-> VOIP Gateway settlement. Calling from Communicator to Gateway users run fine; every call originated by gateway users receive SIP/2.0 400 Invalid Contact Information. Which is the Normalization rule role in inbound OCS calls ? I have a location profile normalization rule: ^\+(\d{5})$  <--> $1  for manage outbond calls, it's checked for inbound calls too ?
  This is the SIP log: (Every attempt with (+) in INVITE/FROM/TO header was done. Same result)
  
  
  INVITE sip:44653@mediation.tsf.local;transport=tcp SIP/2.0
  
  Record-Route: <sip:10.57.3.141:5080;lr;sipX-route=%2Afrom%7EYzE0N2RiMWI%60.authrules%2Aauth%7E%21822274c8a0ce72388ed7b8363600c832>
  
  Max-Forwards: 18
  
  Contact: <sip:+200@10.57.7.20:17216>
  
  To: "+44653"<sip:+44653@10.57.3.141>
  
  From: "+200"<sip:+200@10.57.3.141>;tag=c147db1b
  
  Call-Id: ZjIyODIxOTRjOTM1YjkzMDI3OTllZjAzYjc2Y2ZmMWY.
  
  Cseq: 1 INVITE
  
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  
  Content-Type: application/sdp
  
  User-Agent: X-Lite release 1011s stamp 41150
  
  Content-Length: 412
  
  Date: Wed, 09 Jan 2008 10:26:18 GMT
  
  Via: SIP/2.0/TCP 10.57.3.141:5080;branch=z9hG4bK-a6658b09d29177b46b8edabfd6baf26b
  
  Via: SIP/2.0/UDP 10.57.3.141;branch=z9hG4bK-c737ffd3a2dba9aec9a47ba877f2603c
  
  Via: SIP/2.0/UDP 10.57.7.20:17216;branch=z9hG4bK-d87543-ad437625ba4cde23-1--d87543-;rport=17216
  
  
  
  v=0
  
  o=- 2 2 IN IP4 10.57.7.20
  
  s=CounterPath X-Lite 3.0
  
  c=IN IP4 10.57.7.20
  
  t=0 0
  
  m=audio 5006 RTP/AVP 107 119 100 106 0 105 98 8 101
  
  a=alt:1 1 : UTFJi2Ky +xViCroo 10.57.7.20 5006
  
  a=fmtp:101 0-15
  
  a=rtpmap:107 BV32/16000
  
  a=rtpmap:119 BV32-FEC/16000
  
  a=rtpmap:100 SPEEX/16000
  
  a=rtpmap:106 SPEEX-FEC/16000
  
  a=rtpmap:105 SPEEX-FEC/8000
  
  a=rtpmap:98 iLBC/8000
  
  a=rtpmap:101 telephone-event/8000
  
  a=sendrecv
  
  SIP/2.0 100 Trying
  
  FROM: "+200"<sip:+200@10.57.3.141>;tag=c147db1b
  
  TO: "+44653"<sip:+44653@10.57.3.141>
  
  CSEQ: 1 INVITE
  
  CALL-ID: ZjIyODIxOTRjOTM1YjkzMDI3OTllZjAzYjc2Y2ZmMWY.
  
  VIA: SIP/2.0/TCP 10.57.3.141:5080;branch=z9hG4bK-a6658b09d29177b46b8edabfd6baf26b,SIP/2.0/UDP 10.57.3.141;branch=z9hG4bK-c737ffd3a2dba9aec9a47ba877f2603c,SIP/2.0/UDP 10.57.7.20:17216;branch=z9hG4bK-d87543-ad437625ba4cde23-1--d87543-;rport=17216
  
  CONTENT-LENGTH: 0
  
  
  
  SIP/2.0 400 Invalid Contact information
  
  FROM: "+200"<sip:+200@10.57.3.141>;tag=c147db1b
  
  TO: "+44653"<sip:+44653@10.57.3.141>;tag=34475251be
  
  CSEQ: 1 INVITE
  
  CALL-ID: ZjIyODIxOTRjOTM1YjkzMDI3OTllZjAzYjc2Y2ZmMWY.
  
  VIA: SIP/2.0/TCP 10.57.3.141:5080;branch=z9hG4bK-a6658b09d29177b46b8edabfd6baf26b,SIP/2.0/UDP 10.57.3.141;branch=z9hG4bK-c737ffd3a2dba9aec9a47ba877f2603c,SIP/2.0/UDP 10.57.7.20:17216;branch=z9hG4bK-d87543-ad437625ba4cde23-1--d87543-;rport=17216
  
  CONTENT-LENGTH: 0
  
  SERVER: RTCC/3.0.0.0 MediationServer
  
  
  
  ACK sip:44653@mediation.tsf.local;transport=tcp SIP/2.0
  
  Contact: sip:+200@10.57.7.20:17216
  
  From: "+200"<sip:+200@10.57.3.141>;tag=c147db1b
  
  To: "+44653"<sip:+44653@10.57.3.141>;tag=34475251be
  
  Call-Id: ZjIyODIxOTRjOTM1YjkzMDI3OTllZjAzYjc2Y2ZmMWY.
  
  Cseq: 1 ACK
  
  Max-Forwards: 20
  
  Via: SIP/2.0/TCP 10.57.3.141:5080;branch=z9hG4bK-a6658b09d29177b46b8edabfd6baf26b
  
  Content-Length: 0
  
  
  
  
  

  Wednesday, January 9, 2008 11:10 AM
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  Hi Alberto, has this issue been resolved? I'm experiencing exactly the same issue (posted yesterday) where I can't call from my Mitel IP phone to the Office Communicator client. Had no responses yet, so was wondering if your problem is already solved.

   

  Thanks

   

   

  Friday, January 18, 2008 11:20 AM
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  Issue is still open for me. Maybe SIP/TCP stack of our VOIP Gateway is buggy/lack but I'm not be able to make calls from Dialogic Gateway to OCS mediation server work too.
  

  Monday, January 21, 2008 3:28 PM